VoIP Signaling Protocols
Setting Up and Tearing Down the Call
The first hurdle to overcome when making a VoIP phone call is to establish a connection between the parties involved. In legacy telephony, this is done by switching circuits until a physical wire is established between locations.
The Internet Protocol on the other hand is connectionless by nature. IP packets have a tendency to take whatever route they find first, and end up in whatever order they arrive. For time sensitive applications such as voice and video this is unacceptable. Steps must be taken to establish a point to point connection and to keep it open for the duration of the call.
Similar to the handshake of the DHCP protocol, the VoIP signaling protocols use TCP to set up, manage and tear down the VoIP phone call. Signaling protocols are not concerned with the actual media stream of voice or video, and could care less about QoS and traffic engineering. Their basic functions are to first initiate a session, then to find common ground for communication between the parties involved, and to terminate the session at calls end.
On a closed IP network, real time two way communications between terminals can be managed relatively easily. It is when communications must traverse the WAN, or a switched circuit network e.g. the phone company gets thrown into the mix, signaling protocols must take on additional responsibilities such as address translation, bandwidth management, authorization, and in some cases make routing decisions.
While many sets of protocols have been developed over the years to handle these functions, the ones that have benefited most adhere to the recommendations of the Internet Engineering Task Force (IETF), and/or the ITU – International Telecommunications Union. Standards based protocols promote interoperability between disparate devices, and speed the development process for emerging technologies.
We begin with the H.323 suite of protocols, which came on the scene in the mid nineties, and have grown to be an accepted set of standards for VoIP and Multimedia communications.
H.323
As a recommendation of the ITU-T, H.323 was originally developed for videoconferencing over a packet based network, but was quickly adopted for Voice over IP. As a session layer protocol, its main function is to perform call control and management on an IP network. The H323 specification relies on two additional signaling protocols, H.225 and H.245, for call setup and management.
When a session is initiated between two H.323 devices, the H.225 standard uses the Q931 ISDN protocol to perform setup and teardown functions using TCP for a reliable connection. H.245 then opens another TCP connection to establish the capabilities of the devices, negotiate the codec’s, and determine which ports will be used for the session. A channel is then opened on which the actual media will travel using UDP for transport because of its speed, and relying on the upper layer Real Time Protocol (RTP) for sequencing and timing information.
It is important to note that for session set up, negotiation, and management, TCP is used for its reliability. For time sensitive media such as voice and video, UDP is utilized as the transport mechanism because of its speed and low overhead. The packet size can be further reduced by using RTP header compression, reducing the combined IP/UDP/RTP header from over 65% to as low as 10% of the entire packet size. Smaller means faster, and in IP Telephony, the name of the game is speed.
An H.323 network is comprised of four logical components, not all of which will be needed on every network, and that can reside on a wide variety of devices.
Terminals
A terminal is an endpoint device such as an IP telephone, a computer running an H.323 software application, or a dedicated conferencing device. An H.323 compliant terminal must support H.245 for channel and capabilities negotiation, RAS (Registration, Admission, Status), Q931 for signaling and setup, and support for RTP and RTCP on which to stream the media. Terminals must support audio, with video and T.120 data communications being optional.
Multipoint Control Units
An MCU provides services to allow three or more terminals to participate in a conference. The MCU consists of a Multipoint Controller (MC) and an optional Multipoint Processor (MP). The MC is responsible for H.245 functions (negotiating common ground) while the optional MP handles the actual mixing of media streams, and manages the streams to avoid bandwidth contention. H.323 supports Centralized, Decentralized, and a Hybrid concept of multipoint conferencing. When terminals participating in a conference reside in both a centralized and decentralized environment (mixed) the MCU acts as a bridge between the two.
MC and MP functions can reside on a dedicated component, terminal, a gateway or a gatekeeper, but when endpoints exist off network (i.e. PSTN), it is recommended that the MC be utilized on a gateway.
Gateways
Gateways provide a variety of services, not the least of which is protocol conversion between H.323 networks and non-H.232, e.g. switched circuit networks (SCN). A gateway performs call setup and teardown, translates audio, video and data formats, and can perform RAS for registration with the gatekeeper. On the H.323 side, the gateway uses H.225 and H.245 for call setup and management, and on the circuit switched side, it utilizes the protocols specific to SCNs such as ISDN and SS7. A gateway can be implemented on a gatekeeper, an MCU, or on a voice enabled router or switch.
Gatekeepers
As an optional component that is usually found in larger Enterprise networks, the gatekeeper is the most important component in the H.323 configuration. The gatekeeper manages all the registered terminals, gateways, and MCUs in a single H.323 zone, which can span multiple LAN/WAN segments. Services such as addressing, authorization and authentication of H.323 components, bandwidth management, accounting and billing can all be configured on the gatekeeper. But once they are, all endpoints must obey! The king of the zone can make routing decisions, and can also simplify management of multiple gateways by handling their call control functions in a centralized manner.
While a gatekeeper can be implemented on a gateway or MCU, in larger organizations you will usually find them on a dedicated server (such as Microsoft’s ISA server) or on a Cisco IOS router.
One big advantage that the H.323 standard seems to have over its nearest competitor is in the area of address resolution. The gatekeeper has the ability to use a number of methods and protocols to resolve a destination address. First, it can ask another gatekeeper. If that doesn’t work, it can use Annex G/H.225.0, TRIP, ENUM, or DNS protocols for address resolution
Security enhancements to H.323 are provided by H.235, adding authentication, encryption, and integrity to the mix. Optional password based and PKI security profiles can be used to authenticate the person, and the call signaling channel can be encrypted using TLS or IPSec.
You will find H.323 technology used in many Cisco products, most notably their CallManager application. Microsoft’s NetMeeting also utilizes the H.323 standard, and peer to peer provider Skype offers support for both H.323 and SIP.
Today, another emerging standard has taken the spotlight, and H.323 has come to be considered by some as legacy technology. Nevertheless, what started out as a pioneer in videoconferencing has carved a niche in IP telephony, and is not likely to be supported in the industry solely for backwards compatibility.
The VoIP Signaling Protocols
H.323 | SGCP | MGCP | Megaco-H.248 | SIP
